The telephone CO is a primary component of the communication infrastructure. It is the interconnection point for dedicated access lines and shared network resources (i.e., to and from local residences and businesses) with long distance phone carriers and data network providers. A typical CO will serve between 10,000 and 50,000 customers in the local area immediately surrounding the CO. In large metropolitan areas, the customer base may be as large as 100,000 and, in a few instances, upwards of 200,000 subscribers. Each CO is usually owned and operated by an Incumbent Local Exchange Carrier (“ILEC”), but a Competitive Local Exchange Carrier (“CLEC”), such as an Inter-exchange Carrier (“IXC”), may have access to and be responsible for a significant portion of the CO.
The current telephony infrastructure relies primarily on circuit switching. It is also based upon the following assumptions: (1) bandwidth is a scarce resource; (2) computation is a scarce resource; and (3) terminal devices are simple with limited capabilities in order to provide network security. Under these assumptions, the telephony network needs to manage the bandwidth and computational burden carefully, and intelligence needs to reside inside of the network. Many telecommunications experts, however, have recognized that these assumptions are no longer valid, and have set forth the following new set of assumptions for the communication infrastructure: (1) bandwidth is plentiful and cheap; (2) computation is plentiful and cheap; and (3) terminal devices are “intelligent,” yet capable of providing the necessary security for networks. Under these new assumptions, the main job of the network is to deliver messages between two points and little else, the prime example of which is the Internet.
Considering current trends and looking into the future, it is likely before long most communications will be packet-switched, rather than circuit-switched, and controlled largely from edge devices. One of the main driving forces for this change is the ease of adding new features on the edge instead of inside of the network. With a network that delivers packets under the direction of edge nodes using a simple interface, any reasonably talented programmer could write an application that can communicate with any other application on this network. This will free an incredible amount of creative energy to develop new applications for multi-media, as well as voice. On the other hand, the current telephony network is centrally administered using a protocol for out-of-band control and signaling, known as signaling system 7 (“SS7”) (see, e.g., T.
Russell, Signaling System #7 (2d. Ed.), McGraw-Hill, 1998). SS7 is a closed system, not accessible to, or directly modifiable by, users. It is doubtful that SS7 will be able to adapt to the rich set of features imaginable in an IP-based telephony network, and even if it can be adapted to some of these features, it is unlikely to change quickly enough to realize them.
There is a very strong movement amongst both data and circuit-based networking companies to support IP telephony, or voice over IP (“VoIP”). For instance, some data companies are developing VoIP gateways and gatekeepers, while local and long distance telephony companies are planning on offering VoIP services to residential and commercial customers. The business models in place exploit the cheaper cost of IP hardware and software compared to legacy telephony equipment. In other words, the current driving force of VoIP is the lower cost of the transport medium and switching equipment. IP-based telephony systems, however, can also sound much better than the legacy telephony system and offer many more features. In the end, these features may be a greater source of revenue than the reduced marginal cost of IP equipment.
For example, the current low sound quality of the legacy telephony system is tightly coupled with its Time Division Multiplex (“TDM”) structure. The system samples voice 8,000 times per second, and each sample is represented by 8 bits. This results in the 64 kilobits per second circuits that dominate the current infrastructure. However, this sampling frequency is just one point on the trade-off curve between required data rate and fidelity. Using the well-known sampling theorem, the highest frequency that can be represented with 8,000 samples per second is 4 kHz. Since the human voice can create sounds up to 10 kHz and the human ear can perceive sounds up to 20 kHz, the 4 kHz bandwidth provided by the telephony system is rather small. In other words, as long as the legacy telephone system is used, the full potential of voice quality will never be reached. In contrast, an IP-based telephony system is by no means constrained to 8,000 samples per second. Accordingly, using high-fidelity handsets, CD-quality conversations can easily be supported by IP-based telephony in the near future.
Since the edge nodes of an IP telephony network will support the rapid deployment of new features, it is conceivable that several new enhancements will be made in the areas of voice and multi-media. For example, the IP-based telephony system may facilitate the following voice enhancements: (1) advanced session initiation, such as multi-party calls using bandwidth-efficient IP multicast and automatic party-finding services; (2) simpler computer-telephony integration (“CTI”), such as call centers with web-based, click-to-dial interfaces; (3) advanced emergency functionality, such as the ability to trade off the number of active calls versus voice quality per call, and prioritize emergency systems (i.e., 911) calls using IP differential or integrated services; (4) a distributed signaling architecture that is more reliable, redundant, modular, and inexpensive (i.e., low cost of entry); (5) integrated media such as voice, video, fax, pagers, web, E-mail, and voicemail accessible via a common addressing mechanism and configuration via a common interface; (6) sophisticated and programmable call processing at client devices (i.e., “smart phones”); and (7) end-user encryption and authentication.
Although, in the long run, IP-based telephony may completely replace circuit-switched telephony, in the short run, the legacy telephony system and the IP-based telephony system will most likely co-exist. Hence, the legacy telephony system and the IP-based telephony system need to be able to inter-operate. Accordingly, it is desirable to provide an architecture and design for an IP-based central office (“IPCO”) that provides a framework for supporting both the legacy circuit-switched telephony system and an IP-based telephony system. It is also desirable to provide an IPCO that leverages existing data networking standards and equipment whenever possible, so that only minimal changes to existing networking technologies are necessary in order to achieve the desired functionality.